Receiver intelligibility enhancement system

ABSTRACT

Embodiments of the invention provide a communication device and methods for enhancing audio signals. A first audio signal buffer and a second audio signal buffer are acquired. Thereafter, the second audio signal is processed based on the linear predictive coding coefficients and gains based on noise power of the first audio signal to generate an enhanced second audio signal.

CROSS REFERENCE TO RELATED APPLICATIONS

This application claims the priority date and benefit of and is aContinuation-In-Part Application of a U.S. Non-Provisional applicationSer. No. 12/139,489 entitled ‘Receiver Intelligibility EnhancementSystem’ and filed on Jun. 15, 2008. The entire teachings of the aboveapplication are incorporated herein by reference. This application alsoclaims the benefit of and priority date of provisional patentapplication 60/944,180 filed on Jun. 15, 2007, a parent application ofSer. No. 12/139,489.

FIELD OF THE INVENTION

The invention relates to audio signal processing, and more specifically,the invention relates to systems and methods for enhancing receiverintelligibility.

BACKGROUND

Speech intelligibility is usually expressed as a percentage of words,sentences or phonemes correctly identified by a listener or a group oflisteners. It is an important measure of the effectiveness or adequacyof a communication system or of the ability of people to communicateeffectively in noisy environments. Quality is a subjective measure,which reflects on individual preferences of listeners. The two measuresare not correlated. In fact, it is well known that intelligibility canbe improved if one is willing to sacrifice quality. It is also wellknown that improving the quality of a signal does not necessarilyelevate its intelligibility. On the contrary, quality improvement isusually associated with loss of intelligibility relative to that of thesignal. This is due to distortion that the signal undergoes in theprocess of enhancing it

Communication devices such as mobile phones, headsets, telephones and soforth may be used in vehicles or in other areas where there is often ahigh level of background noise. A high level of local background noisecan make it difficult for a user of the communication device tounderstand the speech being received from the receiving side in thecommunication network. The ability of the user to effectively understandthe speech received from the receiver side is obviously essential and isreferred to as the intelligibility of the received speech.

In the past, the most common solution to overcome the background noisewas to increase the volume at which the speakers of communication deviceoutput speech. One problem with this solution is that the maximum outputsound level that a phone's speaker can generate is limited. Due to theneed to produce cost-competitive cell phones, companies often uselow-cost speakers with limited power handling capabilities. The maximumsound level such phone speakers generate is often insufficient due tohigh local background noise.

Attempts to overcome the local background noise by simply increasing thevolume of the speaker output can also result in overloading the speaker.Overloading the loudspeaker introduces distortion to the speaker outputand further decreases the intelligibility of the outputted speech. Atechnology that increases the intelligibility of speech receivedirrespective of the local background noise level is needed.

Several attempts to improve the intelligibility in communication devicesare known in the related art. The requirements of an intelligent systemcover naturalness of the enhanced signal, short signal delay andcomputational simplicity.

During the past two decades, Linear Predictive Coding (LPC) has becomeone of the most prevalent techniques for speech analysis. In fact, thistechnique is the basis of all the sophisticated algorithms that are usedfor estimating speech parameters, for example, pitch, formants, spectra,vocal tract and low bit representations of speech. The basic principleof linear prediction states that speech can be modeled as the output ofa linear time-varying system excited by either periodic pulses or randomnoise. The most general predictor form in linear prediction is the AutoRegressive Moving Average (ARMA) model where a speech sample of ‘s (n)’is predicted from ‘p’ past predicted speech samples s (n−1), . . . ,s(n−p) with the addition of an excitation signal u(n) according to thefollowing equation 1:

$\begin{matrix}{{s(n)} = {{\sum\limits_{k = 1}^{p}{a_{k}{s\left( {n - i} \right)}}} + {G{\sum\limits_{l = 0}^{q}{b_{l}{u\left( {n - l} \right)}}}}}} & {{Equation}\mspace{14mu} 1}\end{matrix}$

where G is the gain factor for the input speech and a_(k) and b₁ arefilter coefficients. The related transfer function H (z) is given byfollowing equation 2:

$\begin{matrix}{{H(z)} = \frac{S(z)}{U(z)}} & {{Equation}\mspace{14mu} 2}\end{matrix}$

For an all-pole or Autoregressive (AR) model, the transfer functionbecomes as the following equation 3:

$\begin{matrix}{{H(z)} = {\frac{1}{1 - {\sum\limits_{k = 1}^{p}{a_{k}z^{- k}}}} = \frac{1}{A(z)}}} & {{Equation}\mspace{14mu} 3}\end{matrix}$

Estimation of LPC

Two widely used methods for estimating the LP coefficients exist:autocorrelation method and covariance method. Both methods choose the LPcoefficients a_(k) in such a way that the residual energy is minimized.The classical least squares technique is used for this purpose. Amongdifferent variations of LP, the autocorrelation method of linearprediction is the most popular. In this method, a predictor (an FIR oforder m) is determined by minimizing the square of the prediction error,the residual, over an infinite time interval. Popularity of theconventional autocorrelation method of LP is explained by its ability tocompute a stable all-pole model for the speech spectrum, with areasonable computational load, which is accurate enough for mostapplications when presented by a few parameters. The performance of LPin modeling of the speech spectrum can be explained by theautocorrelation function of the all-pole filter, which matches exactlythe autocorrelation of the input signal between 0 and m when theprediction order equals m. The energy in the residual signal isminimized. The residual energy is given by the following equation 4:

$\begin{matrix}{E = {{\sum\limits_{n = {- \infty}}^{\infty}{e^{2}(n)}} = {\sum\limits_{n = {- \infty}}^{\infty}\left( {{s_{N}(n)} - {\sum{a_{k}{s_{N}\left( {n - k} \right)}}}} \right)^{2}}}} & {{Equation}\mspace{14mu} 4}\end{matrix}$

The covariance method is very similar to the autocorrelation method. Thebasic difference is the length of the analysis window. The covariancemethod windows the error signals instead of the original signal. Theenergy E of the windowed error signal is given by following equation 5:

$\begin{matrix}{E = {{\sum\limits_{n = {- \infty}}^{\infty}{e^{2}(n)}} = {\sum\limits_{n = {- \infty}}^{\infty}{{e^{2}(n)}{w(n)}}}}} & {{Equation}\mspace{14mu} 5}\end{matrix}$

Comparing autocorrelation method and covariance method, the covariancemethod is quite general and can be used with no restrictions. The onlyproblem is that of stability of the resulting filter, which is not asevere problem generally. In the autocorrelation method, on the otherhand, the filter is guaranteed to be stable, but the problems ofparameter accuracy can arise because of the necessity of windowing thetime signal. This is usually a problem if the signal is a portion of animpulse response.

Usually in environments with significant local background noise, thesignal received from the receiving side becomes unintelligible due to aphenomenon called masking. There are several kinds of masking, includingbut not limited to, auditory masking, temporal masking, simultaneousmasking and so forth.

Auditory masking is a phenomenon when one sound is affected by thepresence of another sound. Temporal masking is a phenomenon when asudden sound makes other sounds inaudible. Simultaneous masking is theinability of hearing a sound in presence of other sound whose frequencycomponent is very close to desired sound's frequency component.

In light of the above discussion, techniques are desirable for enhancingreceiver intelligibility.

SUMMARY

The present invention provides a communication device and method forenhancing audio signals. The communication device may monitor the localbackground noise in the environment and enhances the receivedcommunication signal in order to make the communication more relaxed. Bymonitoring the ambient or environmental noise in the location in whichthe communication device is operating and applying receiverintelligibility enhancement processing at the appropriate time, it ispossible to significantly improve the intelligibility of the receivedcommunication signal.

In one aspect of the invention, the noise in the background in which thecommunication device is operating is monitored and analyzed.

In another aspect of the invention, the signals from a far-end aremodified based on the characteristics of the background noise at nearend.

In another aspect of the invention, Linear Predictive Coding (LPC)coefficients of a first audio signal buffer acquired from a near-end areused to filter a second audio signal buffer acquired from a far-end togenerate an intelligibility enhanced signal.

BRIEF DESCRIPTION OF THE DRAWINGS

Having thus described the invention in general terms, reference will nowbe made to the accompanying drawings, which are not necessarily drawn toscale, and wherein:

FIG. 1 illustrates an environment where various embodiments of theinvention function;

FIG. 2 illustrates a block diagram of a communication device forenhancing audio signals, in accordance with an embodiment of theinvention;

FIG. 3 is a flow diagram illustrating processing of audio signals, inaccordance with an embodiment of the invention;

FIG. 4 illustrates acquiring and outputting of audio signals by thecommunication device, in accordance with an embodiment of the invention;

FIG. 5 illustrates the communication device as a mobile phone, inaccordance with an embodiment of the invention;

FIG. 6 illustrates the communication device as a headset, in accordancewith an embodiment of the invention;

FIG. 7 illustrates the communication device as a cordless phone, inaccordance with an embodiment of the invention; and

FIG. 8 is a flowchart illustrating enhancing of audio signal, inaccordance with an embodiment of the invention.

DETAILED DESCRIPTION OF THE INVENTION

The following detailed description is directed to certain specificembodiments of the invention. However, the invention can be embodied ina multitude of different ways as defined and covered by the claims andtheir equivalents. In this description, reference is made to thedrawings wherein like parts are designated with like numeralsthroughout. Unless otherwise noted in this specification or in theclaims, all of the terms used in the specification and the claims willhave the meanings normally ascribed to these terms by workers in theart.

The present invention provides a novel and unique technique to improvethe intelligibility in noisy environments experienced in communicationdevices such as a cellular telephone, wireless telephone, cordlesstelephone, and so forth. While the present invention has applicabilityto at least these types of communications devices, the principles of thepresent invention are particularly applicable to all types ofcommunications devices, as well as other devices that process speech innoisy environments such as voice recorders, dictation systems, voicecommand and control systems, and the like. For simplicity, the followingdescription may employ the terms “telephone” or “cellular telephone” asan umbrella term to describe the embodiments of the present invention,but those skilled in the art will appreciate that the use of such termis not to be considered limiting to the scope of the invention, which isset forth by the claims appearing at the end of this description.

FIG. 1 illustrates an environment 100 where various embodiments of theinvention function. A communication device 102 may communicate with afar-end device 108 through a communication channel 112. Examples ofcommunication device 102 and far-end device 108 include, but are notlimited to, a mobile phone, a telephone, a cordless phone, a Bluetoothheadset, a computer, a dictation system, voice recorders and otherdevices capable of communication. Communication channel 112 may be forexample, a wireless channel, a radio channel, a wired channel and soforth. Communication device 102 and far-end device 108 communicate byexchanging signals over communication channel 112. Far-end device 108may be located at a far end 110 from communication device 102, whilecommunication device 102 may be located at a near end 104. Far end 110may be location that is distant from near end 104 of communicationdevice 102. For example, near end 104 may be a restaurant having localbackground noise 106 and far end 110 may be a home or office. Backgroundnoise 106 may be due to talking of other people, machines or devicesused inside or near the restaurant.

Generally in conventional devices the signals received from far-enddevice 108 and outputted through an earpiece of the communication device102 may not sound clear because of the background noise 106. The presentinvention provides techniques to generate and output clear and enhancedsignals from the earpiece of communication device 102.

FIG. 2 illustrates a block diagram of communication device 102 forenhancing audio signals, in accordance with an embodiment of theinvention. Communication device 102 may include multiple microphones 212a-n for acquiring audio signals. The audio signals acquired bymicrophones 212 a-n may be analog and can be converted to digital audiosignals by Analog-To-Digital (ADC) convertors 214 a-n connected tomicrophones 212 a-n. Microphones 212 a-n may acquire audio signals fromnear end 104 of communication device 102. Therefore, the audio signalsacquired by microphones 212 a-n may include background noise. Although,multiple microphones 212 a-n are shown, a person skilled in the art willappreciate that the present invention can function with a singlemicrophone implemented in communication device 102.

A Digital-To-Analog (DAC) convertor 218 connected to an earpiece 216 mayconvert digital audio signals to analog audio signals that may then beoutputted by earpiece 216. Further, communication device 102 includes areceiver 210 that receives signals from a far-end device oncommunication channel 112. An enhancer 202 processes the signalsreceived from microphones 212 a-n and receiver 210 to enhance the signalreceived from receiver 210. Further, the enhanced signal is outputtedfrom earpiece 216. Enhancer 202 may include a processor 204 and a memory206. Processor 204 can be a general purpose fixed point or floatingpoint Digital Signal Processor (DSP), or a specialized DSP (fixed pointor floating point). Examples of processor 204 include, but are notlimited to, processor Texas Instruments (TI) TMS320VC5510, TMS320VC6713,TMS320VC6416; Analog Devices (ADI) BlackFinn (BF) 531, BF532, 533;Cambridge Silicon Radio (CSR) Blue Core 5 Multi-media (BC5-MM) or BlueCore 7 Multi-media BC7-MM and so forth. Memory 206 can be for example, aRandom Access Memory (RAM), SRAM (Static Random Access Memory), a ReadOnly Memory (ROM), a solid state memory, a computer readable media andso forth. Further, memory 206 may be implemented inside or outsidecommunication device 102. Memory 206 may include instructions that canbe executed by processor 204. Further, memory 206 may store data thatmay be used by processor 204. Processor 204 and memory 206 maycommunicate for data transfer through system bus 208.

FIG. 3 is a flow diagram illustrating processing of audio signals, inaccordance with an embodiment of the invention. Background noise 106acquired by microphones 212 a-n may be converted to digital first audiosignal buffer 302. Similarly, audio signals received from far end 110may be processed as second audio signal buffer 310. The audio signalsreceived from far end 110 can be speech signals. In an embodiment of theinvention, background noise 106 and audio signals received from far end110 may be stored as digital first audio signal buffer 302 and secondaudio signal buffer 310 respectively in memory 206 for processing.Further, the contents of first audio signal buffer 302 and second audiosignal buffer 310 may be segmented and windowed for processing. In anembodiment of the invention, the segmentation is done by using a Hanningwindow. However people skilled in the art can appreciate the fact thatthe other windowing schemes, such as Hamming window, Blackman-Harriswindow, trapezoidal window and so forth, can also be used.

At block 304, noise power of first audio signal buffer 302 may becalculated. For example, the noise power can be calculated as shown bypseudo program instructions and equation:

$\begin{matrix}{\mspace{79mu} {{{NoisePower} = 0}\mspace{79mu} {{{Loop}\mspace{14mu} i} = {1\mspace{14mu} {to}\mspace{14mu} P}}\mspace{101mu} {{NoisePower} = {{NoisePower} = {{input}\lbrack i\rbrack}^{2}}}\mspace{79mu} {{End}\mspace{14mu} {loop}}}} & {{Equation}\mspace{14mu} 6}\end{matrix}$

where ‘i’ is an index, ‘P’ is the number of samples in each frame offirst audio signal buffer 302. For example, there can be 160 samples ineach frame for a narrowband communication system. In equation (1),‘input[ ]’ represents first audio signal buffer 302. The result of theabove mentioned instructions is the ‘Noisepower’ of first audio signalbuffer 302. In an embodiment of the invention, the above mentionedinstructions may be stored in memory 206.

Second audio signal buffer 310 is attenuated at a block 314 by a firstgain 313 to generate a third audio signal buffer 318. First gain 313 maybe within a first predefined range. For example the first predefinedrange may be from 10% to 30%. Moreover, second audio signal buffer 310is attenuated at a block 312 by a second gain 315 to generate a fourthaudio signal buffer 316. Second gain 315 may be within a secondpredefined range. For example the second predefined range may be from70% to 90%. Therefore, the sum of first gain 313 and second gain 315 mayequal 100%. The values of gain can be controlled adaptively based on thenose power.

The mean of the noise power (MeanNoisePower) can be calculated by usingequation (2):

$\begin{matrix}{{MeanNoisePower} = \frac{NoisePower}{P}} & {{Equation}\mspace{14mu} 7}\end{matrix}$

Further, Direct Current (DC) components can be removed from first audiosignal buffer 302 as shown by pseudo program instructions and equation:

$\begin{matrix}{\mspace{79mu} {{{{Loop}\mspace{14mu} i} = {1\mspace{14mu} {to}\mspace{14mu} P}}\mspace{101mu} {{{input}\lbrack i\rbrack} = {{{input}\lbrack i\rbrack}^{2} - {MeanNoisePower}}}\mspace{79mu} {{End}\mspace{14mu} {loop}}}} & {{Equation}\mspace{14mu} 8}\end{matrix}$

At block 308, fourth audio signal buffer 316 may be filtered by usingLinear Prediction Coding (LPC) coefficients to generate a fifth audiosignal buffer 322. The LPC coefficients are calculated based on thecomponents of first audio signal buffer 302 after the removal of DCcomponents. In an embodiment of the invention, the LPC coefficients maybe calculated using Durbin-Levinson method. However, people skilled inthe art will appreciate that other techniques such as covariance method,autocorrelation method or other methods may be used to calculate the LPCcoefficients. Thereafter, fifth audio signal buffer 322 is added tothird audio signal buffer 318 at block 320 to generate a sixth audiosignal buffer 324. Sixth audio signal buffer 324 is an enhanced audiosignal that may be converted from digital to analog and outputted fromearpiece 216 of communication device 102. In an embodiment of theinvention first audio signal buffer 302, the second audio signal buffer310, third audio signal buffer 318, fourth audio signal buffer 316,fifth audio signal buffer 322, and sixth audio signal buffer 324 may bestored in memory 206 for processing by processor 204.

FIG. 4 illustrates acquiring and outputting of audio signals bycommunication device 102, in accordance with an embodiment of theinvention. As shown, first audio signal buffer 302 is acquired frommicrophone 212 and second audio signal buffer 310 is received fromfar-end device 108. Communication device 102 transmits signals tofar-end device 108 based on first audio signal buffer 302.

First audio signal buffer 302 and second audio signal buffer 310 areprocessed by enhancer 202 to generate sixth audio signal buffer 324.Sixth audio signal buffer 324 may be converted from digital to analogand outputted from earpiece 216 of communication device 102. Sixth audiosignal buffer 324 is an enhanced form of second audio signal buffer 310that sounds clear to the user of communication device 102 even inpresence of background noise 106.

FIG. 5 illustrates communication device 102 as a mobile phone, inaccordance with an embodiment of the invention. As shown, communicationdevice 102 may include an earpiece 502, a microphone 504, a display 506,a keypad 508, and enhancer 202. Further, mobile phone may communicate toanother device through a mobile network. Microphone 504 acquires firstaudio signal buffer 302 and second audio signal buffer 310 is receivedfrom the other device on the mobile network. Although a singlemicrophone 504 is shown, a person skilled in the art will appreciatethat the mobile phone may include multiple microphones. Enhancer 202processes first audio signal buffer 302 and second audio signal buffer310 to generate an enhanced signal that is outputted from earpiece 502.In an embodiment of the invention, communication device 102 may includea switch (not shown) to activate and/or deactivate enhancer 202.Therefore, once enhancer 202 is deactivated, first audio signal buffer302 and second audio signal buffer 310 are not processed and signalreceived from a far end device is outputted from earpiece 502.

FIG. 6 illustrates communication device 102 as a headset, in accordancewith an embodiment of the invention. Communication device 102 may be aBluetooth headset that can be coupled with a device such as a mobilephone. As shown, the headset may include an earpiece 602, a microphone604 and enhancer 202. Microphone 604 acquires first audio signal buffer302 and second audio signal buffer 310 is received from the other deviceon radio or wireless channel. Although a single microphone 604 is shown,a person skilled in the art will appreciate that the mobile phone mayinclude multiple microphones. Enhancer 202 processes first audio signalbuffer 302 and second audio signal buffer 310 to generate an enhancedsignal that is outputted from earpiece 602. In an embodiment of theinvention, communication device 102 may include a switch (not shown) toactivate and/or deactivate enhancer 202. Therefore, once enhancer 202 isdeactivated, first audio signal buffer 302 and second audio signalbuffer 310 are not processed and signal received from a far end deviceis outputted from earpiece 602.

FIG. 7 illustrates communication device 102 as a cordless phone, inaccordance with an embodiment of the invention. As shown, the cordlessmay include an earpiece 702, a microphone 704, a display 706, a keypad708, an antenna 710 and enhancer 202. The cordless phone may communicatewith a far end device through a docking station (not shown) by usingantenna 710. Microphone 704 acquires first audio signal buffer 302 andsecond audio signal buffer 310 is received from the other device onradio or wireless channel. Although a single microphone 704 is shown, aperson skilled in the art will appreciate that the mobile phone mayinclude multiple microphones. Enhancer 202 processes first audio signalbuffer 302 and second audio signal buffer 310 to generate an enhancedsignal that is outputted from earpiece 702. In an embodiment of theinvention, earpiece 702 may include a loudspeaker.

In an embodiment of the invention, communication device 102 may includea switch (not shown) to activate and/or deactivate enhancer 202.Therefore, once enhancer 202 is deactivated, first audio signal buffer302 and second audio signal buffer 310 are not processed and signalreceived from a far end device is outputted from earpiece 702.

FIG. 8 is a flowchart illustrating enhancing of audio signal, inaccordance with an embodiment of the invention. Communication device 102may communicate with far-end device 108 over communication channel 112.However, communication device 102 may be present at a location havingbackground noise. Therefore, the signals received from far-end device108 may required to be enhanced to make them clear and audible. At step802, first audio signal buffer 302 is acquired from microphones 212 a-nand second audio signal buffer 310 is acquired from far-end device 108.Thereafter, at step 804, the contents of first audio signal buffer 302and second audio signal buffer 310 are segmented. At step 806, thesegmented contents of first audio signal buffer 302 and second audiosignal buffer 310 are windowed. In an embodiment of the invention, thesegmented contents are windowed based on Hanning window. Thereafter, atstep 808, noise power of first audio signal buffer 302 is estimated.Further, a mean noise power may be estimated at step 808. Subsequently,at step 810, first gain 313 and second gain 315 are generated andcontrolled. First gain 313 and second gain 315 are generated based onthe noise power of first audio signal buffer 302. Moreover, first gain313 and second gain 315 can be controlled adaptively based on the noisepower. In an embodiment of the invention, first gain 313 is within afirst predefined range and second gain 315 in within a second predefinedrange. Further, the sum of first gain 313 and second gain 315 equals100%.

Thereafter, at step 812, second audio signal buffer 310 is attenuated byfirst gain 313 to generate third audio signal buffer 318. Further, atstep 814, second audio signal buffer 310 is attenuated second gain 315to generate fourth audio signal buffer 316. In an embodiment of theinvention, steps 812 and 814 may be performed simultaneously. At step816, DC components are removed from first audio signal buffer 302.Thereafter, LPC coefficients of the first audio signal buffer 302 arecalculated. At step 820, fourth audio signal buffer 316 is filteredbased on the LPC coefficients to generate fifth audio signal buffer 322.Subsequently, fifth audio signal buffer 322 is added to third audiosignal buffer 318 to generate sixth audio signal buffer 324. Sixth audiosignal buffer 324 may be converted from digital to analog and outputtedfrom earpiece 216 of communication device 102.

This written description uses examples to disclose the invention,including the best mode, and also to enable any person skilled in theart to practice the invention, including making and using any devices orsystems and performing any incorporated methods. The patentable scopethe invention is defined in the claims, and may include other examplesthat occur to those skilled in the art. Such other examples are intendedto be within the scope of the claims if they have structural elementsthat do not differ from the literal language of the claims, or if theyinclude equivalent structural elements with insubstantial differencesfrom the literal languages of the claims.

1. A communication device for generating enhanced audio signals, thecommunication device comprising: an earpiece configured to output audiosignals; at least one microphone configured to acquire a first audiosignal buffer, wherein the first audio signal buffer comprises a noisesignal; a receiver configured to receive a second audio signal buffer,wherein the second audio signal buffer comprises a speech signalreceived through a communication channel; and a processor configured to:segment the contents of the first audio signal buffer and the secondaudio signal buffer; window the segmented contents of the first audiosignal buffer and the second audio signal buffer; estimate the noisepower of the first audio signal buffer; control a first gain and asecond gain adaptively, wherein the first gain is within a firstpredefined range and the second gain is within a second predefinedrange; attenuate the second audio signal buffer with the first gain togenerate a third audio signal buffer; attenuate the second audio signalbuffer with the second gain to generate a fourth audio signal buffer;remove direct current components from the first audio signal bufferbased on the noise power; calculate Linear Prediction Coding (LPC)coefficients of the first audio signal buffer; filter the fourth audiosignal buffer by using LPC coefficients of the first audio signal bufferto generate a fifth audio signal buffer; and add the fourth audio signalbuffer to the fifth audio signal buffer to generate a sixth audio signalbuffer, wherein the sixth audio signal buffer is outputted by theearpiece.
 2. The communication device of claim 1 further comprising amemory configured to store one or more instructions executable by theprocessor.
 3. The communication device of claim 2, wherein the memory isfurther configured to store one or more of the first audio signalbuffer, the second audio signal buffer, the third audio signal buffer,the fourth audio signal buffer, the fifth audio signal buffer, and thesixth audio signal buffer.
 4. The communication device of claim 1,wherein the processor is further configured to estimate a mean noisepower of the first audio signal buffer.
 5. The communication device ofclaim 4, wherein the processor is configured to remove direct currentcomponents from the first audio signal buffer based on the mean noisepower.
 6. The communication device of claim 1, wherein the processor isconfigured to window the segmented contents of the first audio signalbuffer and the second audio signal buffer by using Hanning window. 7.The communication device of claim 1, wherein the first audio signal isacquired from a near-end of the communication device.
 8. Thecommunication device of claim 1, wherein the communication channelcomprises a wireless communication channel.
 9. The communication deviceof claim 1, wherein the noise power is estimated based on a plurality ofsamples in a plurality of frames of the first audio signal buffer. 10.The communication device of claim 1, wherein the first gain and thesecond gain are controlled based on the noise power.
 11. Thecommunication device of claim 1, wherein the first predefined range isfrom 10% to 30%.
 12. The communication device of claim 1, wherein thesecond predefined range is from 70% to 90%.
 13. The communication deviceof claim 1, wherein the sum of the first gain and the second gain is100%.
 14. A communication device for generating enhanced audio signals,the communication device comprising: an earpiece configured to outputaudio signals; at least one microphone configured to acquire a firstaudio signal buffer, wherein the first audio signal buffer comprises anoise signal; a receiver configured to receive a second audio signalbuffer, wherein the second audio signal buffer comprises a speech signalreceived through a communication channel; and a processor configured to:segment the contents of the first audio signal buffer and the secondaudio signal buffer; window the segmented contents of the first audiosignal buffer and the second audio signal buffer; estimate the noisepower of the first audio signal buffer; control a first gain and asecond gain adaptively based on the noise power, wherein the first gainis in the range from 10% to 30% and the second gain is in the range from70% to 90%, and wherein the sum of the first gain and the second gain is100%; attenuate the second audio signal buffer with the first gain togenerate a third audio signal buffer; attenuate the second audio signalbuffer with the second gain to generate a fourth audio signal buffer;remove direct current components from the first audio signal bufferbased on the noise power; calculate Linear Prediction Coding (LPC)coefficients of the first audio signal buffer; filter the fourth audiosignal buffer by using the LPC coefficients of the first audio signalbuffer to generate a fifth audio signal buffer; and add the fourth audiosignal buffer to the fifth audio signal buffer to generate a sixth audiosignal buffer, wherein the sixth audio signal buffer is outputted by theearpiece; and a memory configured to store one or more instructionsexecutable by the processor.
 15. The communication device of clam 14,wherein the first audio signal is acquired from a near-end of thecommunication device.
 16. The communication device of claim 14, whereinthe memory is further configured to store one or more of the first audiosignal buffer, the second audio signal buffer, the third audio signalbuffer, the fourth audio signal buffer, the fifth audio signal buffer,and the sixth audio signal buffer.
 17. The communication device of claim14, wherein the processor is further configured to estimate a mean noisepower of the first audio signal buffer.
 18. The communication device ofclaim 14, wherein the processor is configured to remove direct currentcomponents from the first audio signal buffer based on the mean noisepower.
 19. A method performed at a communication device for generatingenhanced audio signals, the method comprising: acquiring a first audiosignal buffer, wherein the first audio signal buffer comprises a noisesignal; receiving a second audio signal buffer, wherein the second audiosignal buffer comprises a speech signal received at the communicationdevice through a communication channel; segmenting the contents of thefirst audio signal buffer and the second audio signal buffer; windowingthe segmented contents of the first audio signal buffer and the secondaudio signal buffer; estimating the noise power of the first audiosignal buffer; controlling a first gain and a second gain adaptivelybased on the noise power, wherein the first gain is in the range from10% to 30% and the second gain is in the range from 70% to 90%, andwherein the sum of the first gain and the second gain is 100%;attenuating the second audio signal buffer with the first gain togenerate a third audio signal buffer; attenuating the second audiosignal buffer with the second gain to generate a fourth audio signalbuffer; removing direct current components from the first audio signalbuffer based on the noise power; calculating Linear Prediction Coding(LPC) coefficients of the first audio signal buffer; filtering thefourth audio signal buffer by using the LPC coefficients of the firstaudio signal buffer to generate a fifth audio signal buffer; and addingthe fourth audio signal buffer to the fifth audio signal buffer togenerate a sixth audio signal buffer, wherein the sixth audio signalbuffer is outputted by an earpiece of the communication device.
 20. Themethod claim 19, further comprising estimating a mean noise power of thefirst audio signal buffer, wherein direct current components are removedfrom the first audio signal buffer based on the mean noise power.